The invention relates to a method of transmitting a signal wherein the signal is divided into successive, overlapping blocks by means of windows, the partial signals contained in the blocks then are converted into a spectrum by transformtion, the spectra subsequently are coded, transmitted, decoded after transmission, converted back into partial signals by inverse transformation, and finally the blocks containing the partial signals are joined in an overlapping manner, and wherein the overlapping regions of the blocks are weighted such that the resultant of the window functions in the respective overlapped regions equals 1.
In connection with the transmission of an audio signal, for example, radio transmission, cable transmission, satellite transmission and in connection with recording devices, it is known to convert the analog audio signal into a digital audio signal having a certain resolution to transmit the signal in this form, and to then convert the digital signal back into an analog signal for reproduction. The digital transmission results in a better signal to noise ratio, particularly for reproduction.
The bandwidth required for the transmission of such a signal is essentially determined by the number of sampling values to be transmitted per unit time and by the resolution.
In practice, the requirement exists to keep the required transmission bandwidth as small as possible so as to be able to use a narrowband channel or to transmit as many audio signals as possible simultaneously over an existing channel. The required bandwidth can be reduced by a reduction of the sampling values or the number of bits per sampling value.
However, the consequence of this measure is, as a rule, a deterioration of the reproduction. In a known method (DE OS 35 06 912.0) the reproduction quality is improved in that the digital audio signal is transformed in successive time segments into a short-time spectrum which represents the spectral components of the signal for the respective time segment of, for example, 20 ms. On the basis of psycho-acoustic laws, components which are not perceived by the listener, that is, those which are irrelevant for communications purposes, can generally be found more easily in the short-time spectrum than in the time domain. These components are given less weight or are omitted entirely in transmission. With this measure, a considerable portion of the otherwise necessary data can be omitted from the transmission so that the average bit rate can be markedly reduced.
A method described by J.P. Princen and A.B. Bradley in an article, entitled "Analysis/Synthesis Filter Bank Design Based On Time Domain Aliasing Cancellaticn", in IEEE Transactions on Acoustics, Speech and Signal Processing, Volume ASSP-34, pages 1153 to 1161, October, 1986, is suitable for dividing the signal into segments. Here a transformation is described in which overlapping blocks are produced in the frequency domain with rounded-off window functions in the windows without additional coefficients. In this method, N values are first cut out of the input signal with the aid of a window function f(n) of a length N, an&lt;then transformed into N/2 significant coefficients in the frequency domain. The retransformation calculates N sampling values from the N/2 coefficients which are in turn weighted with the window function f(n).
However, the output signal of the retransformation differs from the input signal of the original transformation. The exact reconstruction of the input signal becomes possible only in that the output values of successive retransformations are added in the overlapping region of every N/2 sampling values. In order to recover. The input signal by means of this so-called "overlap-add" method, the window function f(n) must meet the following conditions: ##EQU1##
The first condition corresponds to symmetry of f(n). The second condition corresponds to point symmetry of the square of f(n) in each window half. Under consideration of these conditions, the effective window length of the transformation can be varied between N/2 and N sampling values.
When using these methods in transformation coders, the selection of the window length leads to the following consequences. A long window length with, preferably, a rounded-off form permits good frequency selectively. Thus, the retransformation, the error is expanded over the entire effective window length due to the quantization of the coefficients. This can have a negative effect on the subjective quality of the coded signal, particularly with large jumps in the amplitude of the signal to be coded.
The selection of shorter windows causes a deterioration of the frequency selectivity, which has a negative effect on the transformation gain to be attained, particularly with strongly correlated input signals. On the other hand, the errors can be limited to the respective window by quantization of the coefficients during large signal jumps so that their affecting the adjacent windows can be prevented.